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https://github.com/mollyim/webrtc.git
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d5ebc33562
NetEq packet source input doesn't have any other uses than rtp dump, so remove that layer. Bug: None Change-Id: I667bb4aead9f0f2fe8a1c0d6d911a4670ded67e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300542 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39810}
109 lines
3.5 KiB
C++
109 lines
3.5 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h"
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#include "absl/strings/string_view.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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namespace webrtc {
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namespace test {
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namespace {
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// An adapter class to dress up a PacketSource object as a NetEqInput.
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class NetEqRtpDumpInput : public NetEqInput {
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public:
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NetEqRtpDumpInput(absl::string_view file_name,
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const std::map<int, RTPExtensionType>& hdr_ext_map,
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absl::optional<uint32_t> ssrc_filter)
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: source_(RtpFileSource::Create(file_name, ssrc_filter)) {
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for (const auto& ext_pair : hdr_ext_map) {
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source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
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}
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LoadNextPacket();
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}
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absl::optional<int64_t> NextOutputEventTime() const override {
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return next_output_event_ms_;
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}
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absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
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return absl::nullopt;
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}
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void AdvanceOutputEvent() override {
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if (next_output_event_ms_) {
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*next_output_event_ms_ += kOutputPeriodMs;
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}
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if (!NextPacketTime()) {
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next_output_event_ms_ = absl::nullopt;
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}
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}
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void AdvanceSetMinimumDelay() override {}
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absl::optional<int64_t> NextPacketTime() const override {
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return packet_ ? absl::optional<int64_t>(
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static_cast<int64_t>(packet_->time_ms()))
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: absl::nullopt;
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}
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std::unique_ptr<PacketData> PopPacket() override {
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if (!packet_) {
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return std::unique_ptr<PacketData>();
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}
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std::unique_ptr<PacketData> packet_data(new PacketData);
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packet_data->header = packet_->header();
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if (packet_->payload_length_bytes() == 0 &&
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packet_->virtual_payload_length_bytes() > 0) {
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// This is a header-only "dummy" packet. Set the payload to all zeros,
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// with length according to the virtual length.
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packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
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std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
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} else {
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packet_data->payload.SetData(packet_->payload(),
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packet_->payload_length_bytes());
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}
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packet_data->time_ms = packet_->time_ms();
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LoadNextPacket();
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return packet_data;
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}
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absl::optional<RTPHeader> NextHeader() const override {
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return packet_ ? absl::optional<RTPHeader>(packet_->header())
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: absl::nullopt;
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}
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bool ended() const override { return !next_output_event_ms_; }
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private:
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void LoadNextPacket() { packet_ = source_->NextPacket(); }
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absl::optional<int64_t> next_output_event_ms_ = 0;
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static constexpr int64_t kOutputPeriodMs = 10;
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std::unique_ptr<RtpFileSource> source_;
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std::unique_ptr<Packet> packet_;
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};
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} // namespace
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std::unique_ptr<NetEqInput> CreateNetEqRtpDumpInput(
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absl::string_view file_name,
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const std::map<int, RTPExtensionType>& hdr_ext_map,
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absl::optional<uint32_t> ssrc_filter) {
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return std::make_unique<NetEqRtpDumpInput>(file_name, hdr_ext_map,
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ssrc_filter);
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}
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} // namespace test
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} // namespace webrtc
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